Please keep any discussion of the test sample in this tread, rather than where it's simply "stored".

The problem:If oversampled the true peak is reveal to be almost +11dB.A DAC would need 11dB headroom (or alternatively ~12dB which equals 2 bits) to handle this wav correctly.

The solution?:A "quick fix" for a 24bit (or float) audio chain, would be to reduce the volume by 12dB somewhere.Volume loss can be later compensated by simply increasing the analog volume (the user turning the knob a little higher).

*** The rest is somewhat opinionated. ***

Thoughts:As such the "bottom 3 bits" of audio could be considered waste-able, 2 bits to handle pathological intersample peaks, + 1 bit due to quantization/noisefloor/dither.A "24bit" DAC would have no issues, 21bits to use is a lot. Likewise a "20bit" DAC would still have 17bits to use.Ideally the 11 (or 12) dB volume reduction would be done by the DAC just before the reconstruction stage.

Issues?:For a 12dB headroom DAC one would need to crank up the playback volume, so such a DAC would sound more quiet than most other DACs.Noisefloor of the amplifier and other parts of the equipment/audio chain is also an issue.But even "cheap" gear has around -80dBFS to -100dBFS noisefloor.Also considering that a normal living room can easily have a +50dB noisefloor, so loosing out on the 12dB or so of the quietest audio is not an issue.

So if taking CD audio as an example, a 12dB adjustment would cause the content in the -96dBFS to -84dBFS range to be lost.The loss can be avoided by simply passing the 16bit audio as 24bit or 32bit float instead.Under Windows Vista and Windows 7 and Windows 8 all audio is changed to 32bit so this is a non-issue.

How to avoid intersample peaks on gear without the needed headroom?:On Windows you can simply make sure that you never raise the volume (in Windows) above -12dBFS (~45% volume),and instead use the analog volume knob (if there is one on your system or gear) instead.

11dB really?Yep! Then again this is a pathological example."Normally" the intersample peak is within 1dB of the digital peak, and in some rare cases up to 2 to 3dB higher.If you make/master music, then the final mix/pressing master/encoding/exporting should have 2 or 3dB headroom.So as long as no peaks go above 3dB you should be pretty darn safe from causing any clicks or distortion for the end user.

The example here is a pure spike, and humans tend not to like to listen to pops, clicks, static, test tones, or similar.So encountering anything like this "in the wild" is very rare.

Is it really that bad?:Please remember that intersample peaks do not damage equipment, at least I've never heard or read about such happening, and the CD was invented like ages ago.So if this was a practical issue we'd have heard about it along time ago as equipment got fried etc. And we'd have had a solution years ago as well.

The only thing it does is damage the audio quality, that is if you actually can hear/notice it at all. You are more likely to hear crackling/distortion from overly compressed music.And ironically it is that type of overly compressed music that has the most intersample peaks that go above 0dBFS.Solution? Stop compressing the hell out of music. Use 20dB or more headroom and intersample peaks will most likely never be an issue.

When designing Benchmark's new DAC2 HGC D/A converter, we chose to add 3.5 dB of digital headroom to accommodate inter-sample overs. We are working with a 32-bit fixed-point conversion system, and a 32-bit fixed-point gain control. The conversions subsystem has a 133 dB SNR, so we can afford to throw away 3.5 dB SNR to eliminate the clipping of inter-sample overs.

A survey of our in-house music library showed inter-sample overs reaching peak levels of +1.5 to +2 dBFS worst-case. However, please note that our entire library is ripped in lossless formats. I suspect that inter-sample overs could be higher in amplitude, and more frequent, when the audio is reconstructed with an MP3 decoder. Does anyone have test results for MP3 audio sources?

I believe 3.5 dB of headroom above 0 dBFS is sufficient to handle all continuous waveforms, including square waves. Can anyone provide examples or calculations to prove otherwise?

3.5 dB of headroom should also be more than sufficient to handle music (but my tests are limited to lossless rips at standard sample rates between 44.1 and 192 kHz).

The example cited in this thread is high-amplitude high-frequency transient - something we are unlikely to see in a typical recording. It should not be necessary to provide the full 11 dB of headroom required for this pathological example.

A survey of our in-house music library showed inter-sample overs reaching peak levels of +1.5 to +2 dBFS worst-case. However, please note that our entire library is ripped in lossless formats. I suspect that inter-sample overs could be higher in amplitude, and more frequent, when the audio is reconstructed with an MP3 decoder. Does anyone have test results for MP3 audio sources?

I believe 3.5 dB of headroom above 0 dBFS is sufficient to handle all continuous waveforms, including square waves. Can anyone provide examples or calculations to prove otherwise?

3.5 dB of headroom should also be more than sufficient to handle music (but my tests are limited to lossless rips at standard sample rates between 44.1 and 192 kHz).

I'm seeing similar mentioned elsewhere too, normally you would never see above 3. Usually the same as or +1 to +2, very rarely +3, and above +3 probably almost ever, so 3.5 is a good margin. Anything above that are either test/synthetic like my test sample, or a single (or similar) pop that a ed user rarely hears. (usually data corruption during transmission) Although vinyl (being not just analog but mechanical) could cause high intersample peaks by mistake. (vinyl music is usually 40Hz-16KHz)

I'm curious of raw number tests as well, but I can't find a R128 scanner with True Peak with log generation for later processing.At least I have not found such a tool, I'd be happy to scan and provide the results as I'm sure others would be.

I tried with Sox but upsampling (if it's the correct way, or is rate the better option?) even if it's correct, the stats option seem to clip the peak at full signal, and there seem to be no way to change that, using the vol option to reduce by 12dB and then do the upsample does make things better, but -5.20dB (and thyen calculating +12dB) it is still nowhere close to the actual 10.87dB.Shame as the stats that Sox output is otherwise pretty nice.